[Asterisk-Dev] NO DTMF in the Outgoing call

[Chris Heiser]

Areski,

I have had similar problems especially when attempting to send DTMF to a
Cisco ATA186 attached to a PBX.  Attached is a patch for rtp.c to increase
the duration of RFC2833 DTMF tones to a more reasonable value.  Try it out
and see what happens!

--Chris

> -----Original Message-----
> From: asterisk-dev-admin at lists.digium.com [mailto:asterisk-dev-
> admin at lists.digium.com] On Behalf Of Areski
> Sent: Wednesday, February 11, 2004 10:50 AM
> To: asterisk-dev
> Subject: [Asterisk-Dev] NO DTMF in the Outgoing call
> 
> Hello all,
> 
> I cannot have DTMF working when I m making an outgoing call from Asterisk.
> I tried on different servers with different  Asterisk versions.
> I tested all configurations possible with X-lites and also with a
> CiscoAS5300,
> no way... always the same problem.
> 
> I was trying to do some modifications in chan_sip but I m pretty novice
> with Asterisk sources and I didn't find out anything.
> After scanning the sip log, I m thinking it's perhaps an RTP problem.
> Below, I attached the sip log...
> 
> I will greatly appreciate if someone can give me some advices.
> Kind regards,
> Areski
> 
> 
> 
> ------------- SIP DEBUG OUTGOING-CALL : ASTERISK -> X-LITE ----------
> 
> 
> We're at 192.168.1.252 port 16642
> Answering with preferred capability 8
> Answering with non-codec capability 1
> 12 headers, 9 lines
> Reliably Transmitting:
> INVITE sip:phone1 at 192.168.1.29 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> To: <sip:phone1 at 192.168.1.29>
> Contact: <sip:asterisk at 192.168.1.252>
> Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Wed, 11 Feb 2004 15:42:36 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 191
> 
> v=0
> o=root 11215 11215 IN IP4 192.168.1.252
> s=session
> c=IN IP4 192.168.1.252
> t=0 0
> m=audio 16642 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>  (no NAT) to 192.168.1.29:5060
> 
> 
> Sip read:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> Contact: <sip:phone1 at 192.168.1.29:5060>
> Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> CSeq: 102 INVITE
> User-Agent: X-Lite build 1079
> Content-Length: 0
> 
> 
> Sip read:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> Contact: <sip:phone1 at 192.168.1.29:5060>
> Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> CSeq: 102 INVITE
> User-Agent: X-Lite build 1079
> Content-Length: 0
> 
> 
> Sip read:
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> Contact: <sip:phone1 at 192.168.1.29:5060>
> Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> CSeq: 102 INVITE
> Content-Type: application/sdp
> User-Agent: X-Lite build 1079
> Content-Length: 197
> 
> v=0
> o=phone1 161667652 161667652 IN IP4 192.168.1.29
> s=X-Lite
> c=IN IP4 192.168.1.29
> t=0 0
> m=audio 8000 RTP/AVP 8 101
> a=rtpmap:8 pcma/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 
> 10 headers, 9 lines
> Found audio format ALAW
> Found audio format UNKN
> Found description format pcma
> Found description format telephone-event
> Capabilities: us - 8, them - 8/0, combined - 8
> Non-codec capabilities: us - 1, them - 1, combined - 1
> list_route: hop: <sip:phone1 at 192.168.1.29:5060>
> set_destination: Parsing <sip:phone1 at 192.168.1.29:5060> for address/port
> to send to
> set_destination: set destination to 192.168.1.29, port 5060
> Transmitting:
> ACK sip:phone1 at 192.168.1.29:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.252:5060;branch=z9hG4bK708ccbe6
> From: "asterisk" <sip:asterisk at 192.168.1.252>;tag=as3c490458
> To: <sip:phone1 at 192.168.1.29>;tag=as3c490458
> Contact: <sip:asterisk at 192.168.1.252>
> Call-ID: 4846190e0bf039bf637a5cc4442ba507 at 192.168.1.252
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
> 
>  (no NAT) to 192.168.1.29:5060
> 
> 
> 
> _______________________________________________
> Asterisk-Dev mailing list
> Asterisk-Dev at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-dev
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
> 

-------------- next part --------------
A non-text attachment was scrubbed...
Name: rtp.c.patch
Type: application/octet-stream
Size: 1591 bytes
Desc: not available
Url : http://lists.digium.com/pipermail/asterisk-dev/attachments/20040211/b1cf4b81/rtp.c.obj